The present invention relates to telecommunication network analysis, and more particularly to determination of speech latency across a telecommunication network element.
In a third generation (3G) telecommunication scenario “packet based” networks, such as UMTS (Universal Mobile Telecommunication System), permit an optimized use of band resources, adopting Adaptive Multi Rate (AMR) codecs for speech compression and Discontinuous Transmission (DTX) techniques to meet customer satisfaction as a trade-off between Quality of Service (QoS) and costs. The QoS provided by such telecommunication networks depends upon a number of factors, including an overall end-to-end speech latency and distortion introduced using low bit rate codecs.
Speech latency is a time delay between a speech signal at the input of a network device and the same signal at its output, e.g., across two sides of a media gateway in the UMTS architecture. This delay depends on propagation time of the speech signal through the network device, on buffering mechanisms used by codecs (typically at least 20 ms of data packets are buffered before starting an encoding algorithm), and on processing time spent by transcoding equipment for encoding/decoding and forwarding data packets. Moreover, speech latency over UMTS interfaces is affected by typical phenomena occurring in “packet based” networks, such as jitter and packet loss.
Jitter is a packet delay variation due to non-constant arrival times of data packets. The effect of this phenomenon may be attenuated using de-jitter buffers, but such buffers introduce further end-to-end delay. Adaptive de-jitter buffers also may be used with a variable length modified as a function of the monitored jitter during speech pauses. Nevertheless, such adaptive de-jitter buffers introduce a variable speech latency that may impact the overall quality perceived by customers.
Packet loss and packet duplication influence QoS in terms of distortion and speech latency, especially in the presence of burst packet loss when consecutive packets are not received at the network end points. To reduce the effect of this impairment, ad-hoc packet loss concealment (PLC) techniques commonly are used which reduce the distortion perceived by listeners and allow correct reconstruction of the speech signal envelope in the time domain. The disadvantage of these techniques is the requirement of a precise packet loss evaluation, i.e., check of the frame number field for consecutive packets, which cannot be guaranteed in the early stages of the design of 3G network elements.
Currently the technical problem of the assessment of speech latency in a telecommunication network is solved by using an “end-to-end” approach. Manufacturers and operators set up calls between two handsets, saving digital speech signals at the two termination points, i.e., talker mouth and listener ear, and comparing them using end-to-end off-line algorithms. However, even if this approach allows the evaluation of the overall speech latency between two terminating points, it does not provide an accurate measure of the delay introduced by each current or new digital network element within the network. At the same time the use of low bit rate codecs, such as the AMR codecs, no longer allows for predicting theoretically and with high precision the part of the overall delay due to encoding algorithms. This particularly applies to codecs with no linear speech-dependent complexity.
What is desired is the determination of speech latency across a communication network element as opposed to end-to-end speech latency.